If you are leaking file descriptors, doing do a bash ls on /proc/nnn/fd, where nnn is the PID of Asterisk, should not result in a continual increase in the number of pseudo files listed. If you are leaking some other resource, you will have to do an appropriate measurement (ps -l will tell you if you are leaking memory).
A resource leak would indicate a bug, in which case you may have to upgrade Asterisk or submit a but report and wait for it to be fixed, depending on whether it is a known or unknown bug.
Statistics : Posted by kamrul.khan • on Fri Jan 03, 2014 1:08 am • Replies 5 • Views 183
Thanks for your reply. We have updated the ulimit. Initially it was set to 1024. We updated it to 65536. But still we are facing the same issue. We got the below error from the log during the time of occurrence.
ulimit -a gives
We are really in trouble with this issue. Please help us. Thanks in advance.
Statistics : Posted by kamrul.khan • on Fri Jan 03, 2014 1:08 am • Replies 5 • Views 183
I assume you have run out of some system resource. I'd check you don't have any effective limit on open file descriptors, but otherwise you will need to determine if asterisk -r is crashing, in which case you will need a backtrace, or you will need to do a system call trace to get an idea of where it is failing.
Statistics : Posted by kamrul.khan • on Fri Jan 03, 2014 1:08 am • Replies 5 • Views 183
We have this problem in our Asterisk server. When number of active calls reaches around 140 we can no more connect to asterisk console. asterisk -r only shows the welcome message and then exits to shell automatically.
We have Intel E3-1245 4 core processor with 32 GB RAM. CPU usages goes around 40-42% when this kind of situation arises. We also found that the server doesnt actually drop any calls but we simply cant monitor our server when this kind of situation arises.
Please give us some clue like what can be the cause of this problem and how can we fix it. Let us know if you need any further details.
Thanks in advanced!
Statistics : Posted by kamrul.khan • on Fri Jan 03, 2014 1:08 am • Replies 5 • Views 183
I am trying to edit system.conf via AMI UpdateConfig command. But since /etc/dahdi/system.conf does not have standard configuration file format as others (eg. extens.conf, chan_dahdi.conf), AMI always throws back this error to me. Any idea how it can be resolved, or am I trying something which is not supported.
Statistics : Posted by kartus517 • on Fri Jan 10, 2014 2:25 pm • Replies 1 • Views 33
Setup: Asterisk 11 server on Centos 6.4 64bit, and it runs fine. And I'm new to Asterisk.
Calling between internal extensions works fine and presents no issues. But there is an issue with calls to/from the outside world, in that the status of the call (answered, declined) doesn't seem to be passed along. For example, if I call from x1001 to 555-555-5555, the call can be answered with working audio and speech but it dies after 30 seconds. And if the call is declined, the server still attempts to connect the call until the 30-second mark is reached (ie - even if declined, the server will continue to ring the line).
I understand that the 30 seconds mark is from a default that is setting the amount of time to attempt a call. The real issue I am trying to determine is why the status of the call (answered, declined) isn't being transmitted to/from the outside world. I am 99.99% sure it is a simple server configuration issue that I have overlooked, but I don't know exactly what needs changed. Obviously a critical response of some sort is not being passed along.
To bridge from the Asterisk server to the outside world, I am using Twilio.
Sip.conf:
Extensions.conf:
Here is the CLI output during a call, that is answered, from x1001 to 555-555-5555. (Yes, I am using a real phone number when testing this, but I'm not posting it here, of course.). The output is the same whether the call is answered or declined.
Traffic captures of a call from x1001 to 555-555-5555 that is answered or declined looks identical. In the following I've made the data anonymous, so here is the legend:
1.) 111.111.111.111 is the IP address of my Asterisk server, which is my.asterisk.server.com 2.) 222.222.222.222 is the IP address of the xlite1 client 3.) 107.21.222.153 is the IP address of the Twilio SIP server passing the information
And here is a traffic capture of a call between xlite1 (x1001) and xlite2 (x1002). As I said before, these give me no issues. But I'm providing them for trouble shooting purposes in case it helps any of you answer my question.
Statistics : Posted by pk71p • on Sat Jan 11, 2014 9:02 pm • Replies 1 • Views 58
So this is LAN to LAN, your first RTP debug show a Public IP, you need to fix that. Check your Asterisk peer's settings about NAT and in the JsSIP API check the ICE or use the workaorund, setting ICE server to null.
Install the libraries, re run the configure script and recompile asterisk.
You debug show a Public IP and you need to provide both logs, so try first fixing the NAT/ICE stuff.
Statistics : Posted by wacky • on Sun Jan 12, 2014 11:59 am • Replies 3 • Views 74
1) The correct IP negotiation in the SDP of both Asterisk and JsSIP
2) Validate that your Asterisk is compiled with the uuid-devel/libuuid-devel libraries in order to enable ICE(your rtp doesnt show the label "via ICE").
3) If you are in the same LAN as the PBX check that the RTP audio goes to the LAN IP not to the Public IP, use the workaround in the JsSIP PI setting the STUN server to 'null'.
4) Provide both sip debug logs from JsSIP and Asterisk.
Statistics : Posted by wacky • on Sun Jan 12, 2014 11:59 am • Replies 3 • Views 74
he h is the standard "hang-up" extension. The h extension, if it is configured, is called when a caller hangs up the phone. Note that as soon as this happens, the content of ${EXTEN} changes to h.
When the 'h' extension is running, the call legs have already been torn down. There is no way to delay this happening, and you can't do anything in the 'h' extension that needs to read audio from the channel (since no audio will appear, the first time it tries to read audio it will abort). Thus Playback() or Background(), for example, does not work. Essentially, the only things that make sense to use in the 'h' extension are those that don't have anything to do with the external channel that was involved before the hangup. No audio, no DTMF, etc.
Statistics : Posted by batteredveg • on Sun Jan 12, 2014 12:15 pm • Replies 1 • Views 32
I've got what is (hopefully) a simple query; I know what code I need to use, but unsure of where I can put it to actually work.
At the end of every call (whether internal or external), I want a sound file to play to whoever is on the line after the other party hangs up. The sound file simply repeats "The other person has hung up" twice, then hangs up; very similar to how our PSTN lines work in the UK.
Whereabouts can I put this into my FreePBX (or custom) dial plan to work?
Thanks so much
Paul
Statistics : Posted by batteredveg • on Sun Jan 12, 2014 12:15 pm • Replies 1 • Views 32
I'm having troubles with a phone. We have Cisco spa525G handsets, about 8 of them. All inside the network and two outside but those two are connected through an IPsec tunnel that basically extends the network remotely to that site.
We do have one other phone that is outside the network, and did originally work but then the power went out to the main office where the phone server is, and also to the client house, where the external phone is.
From what the phone configs look like, it seems that it was pointing at the external side of our router and should be pushed to the phone server. However it doesn't seem to be able to. The internet is fine and the phone works on the internal network. I've checked both router configs and they seems fine.
Any help would be greatly appreciated. Just let me know if you need anymore information.
Thanks heaps
Daniel
Statistics : Posted by dmoyle • on Sun Jan 12, 2014 5:04 pm • Replies 0 • Views 16
Hi, I'm using asterisk 11, I just want to build a call center setup.for that I made custom queue of having only one extension and custom recording for MOH and I put that .wav file in /var/lib/asterisk/mohmp3/vinc.....and I called that in /etc/asterisk/musiconhold.conf..........and called in my queue as music=vinc..............
for testing I made a call, I heard that customised MOH recording in my headset while queue was (ringing voice) ringing, I answered the call. At the same time I jz made another call to that queue to test the waiting tone ....the call was on wait (ringinuse=disabled) and is also playing same MOH with call position and all in Queue. actually what I want is to separate the 3 instances.
same custom recording for waiting time and Hold time in Queue and differentiate ringing voice of Queue as default "tring tring" or else.
I tried setmusiconhold in Dialplan .........these 3 instances are getting same MOH. Is there any possibility to separate these ?????
Thanks in advance ....
Statistics : Posted by vinc • on Fri Jan 10, 2014 5:22 am • Replies 1 • Views 51
We'd like to be able to have one person on their computer be on multiple phone calls at once. Is this something asterisks can do? Some features we'd want.
Calls start/end at different times Call recordings Sound and mic for each call separately (can talk on one while the other is muted) Ideally play audio files (automated responses).
Thanks.
Statistics : Posted by ryandetzel • on Mon Jan 13, 2014 3:26 pm • Replies 0 • Views 19
